The invention relates to a method for multi-channel processing in a multi-channel sound system, particularly using surround multi-channel sound technology, having the following method steps:                a) splitting a channel or a channel mixture into individual channels;        b) processing the resulting individual channels by means of setting the parameter channel fader;        c) limiting the individual channels by means of setting the values of the parameters channel fader, threshold, release, and output level; and        d) encoding the individual channels.        
The development of multi-channel sound systems has particularly been driven forward by the Dolby laboratories. Tied in with the “Dolby-Surround” invented in the 70s by the Dolby laboratories, in the meantime so-called “matrix surround methods” exist, such as, for example, Dolby ProLogic, ProLogic 2, Circle Surround, Circle Surround 2. In this way, the possibility is created of encoding (coding) up to 7.1 channels, i.e. the channels FL (FL=front left), C (C=center), FR (FR=front right), the side surround channels LS (left surround), RS (right surround), the back surround channels BL (back left), BR (back right), as well as the channel LFE (low frequency effect). From these channels, two transmission channels Lt (L=left, t=total), Rt (R=right, t=total) are matrixed, which contain all the data so that they can be distributed to the original channels again, by corresponding decoders, i.e. reproduced as the original channels after decoding. In the encoding (coding), the audio components of the data that lie in the surround channels LS, RS, BL, BR, are added to the channels L and R with a phase rotation by +/−90°, and embedded in the front channels R, L with slight lowering of the volume level. This encoding of the individual channels is preceded by further method steps in multi-channel sound processing: A first step in the usual multi-channel processing process, in the sector of upmixing of audio material from stereo/mono to surround of any configuration, is splitting a channel or a channel mixture into individual channels. This splitting can be implemented by means of corresponding software. After splitting, the individual channels are available for further processing, particularly in order to guarantee the stability of the multi-channel mix in comparison with the original, for example stereo mix, by means of multi-channel compressing/limiting. For this purpose, a compressor/limiter is provided. A compressor/limiter is a limiter that prevents a peak level from being exceeded, in order to prevent overloads. Also, they can be understood as volume/loudness regulators of instrument groups/singing or speech. Furthermore, they compensate part of the energy loss that results from splitting up a channel or a channel mixture into the individual channels, even before encoding. In this connection, a compressor/limiter is assigned to each individual channel or channel groups. The ratio of the individual channels relative to one another is regulated by means of setting the values of the parameters channel fader, threshold, attack (if available), release, as well as output level (output level) within a compressor or/and limiter. The regulation has the result that the volume/loudness balance of the individual channels relative to one another is already stabilized before encoding. The channel fader volume serves to regulate the volume of each individual channel in a mixing console, which is an important component of a sound studio. The normal volume setting of a channel fader is 0 dB. If one sets a channel fader to 0 dB, then the signal originally applied to the individual channel sounds exactly the same way as it was originally adjusted, presuming a correctly measured and neutrally set gain value at the corresponding channel strip. The channel gain value regulates the pre-amplification of a signal before it passes through the channel fader. The threshold value acts like a threshold of the signal that is applied to the channel fader. In this connection, a compressor/limiter works in such a manner that the compressor/limiter limits the signal as soon as the applied signal exceeds the threshold value. The release value gives information about the time that the compressor/limiter needs to be brought to the zero position again, after the applied signal has dropped below the threshold value again. The attack value determines the reaction time when the threshold value is exceeded. Finally, the output level indicates how strong the signal applied to the channel is after processing by the compressor/limiter. The output leveler is essentially a signal amplifier.
The method of the type indicated initially finds hardly any or only little acceptance within the scope of the matrix surround technology outlined here. It is true that it is confirmed in the relevant technical literature that the matrix surround technology cannot keep up with today's discrete digital methods (see, for example, Christian Birkner, “Surround, Einführung in die Mehrkanaltontechnik [Surround, introduction to multi-channel sound technology], PPV Presse Projekt Verlags GmbH [publisher], Bergkirchen, 2002).
This results, among other things, from the recognition that the Lt, Rt stereo sum created according to the common technical standards, seen in and of itself, cannot keep up qualitatively in comparison with the conventionally produced stereo mixes such as those that are generally processed at one hundred percent within the programs, in the TV sector, radio sector, and music sector. The phase rotations that are caused by the encoding weaken the sound and influence the frequency response, so that they sound “smaller and spongier.”
On the other hand, the matrix surround technology fulfills all the demands on a usable compatible surround system.